February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Question regarding "uuid_bridge needs at least one leg to be answered"
by dma 15 Feb '12

15 Feb '12

Hi All, On the wiki page for uuid_bridge, there is following description: -------------------------------------- uuid_bridge Bridge two call legs together. Usage: uuid_bridge <uuid> <other_uuid> uuid_bridge needs atleast any one leg to be answered. -------------------------------------- Here, "uuid_bridge needs at least any one leg to be answered". Does this limit also apply to dialplan bridge and/or session:bridge in lua script? Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Question-regarding-uuid-bridg… Sent from the freeswitch-users mailing list archive at Nabble.com.

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needs some advice to secure my system
by Philippe Le Toquin 15 Feb '12

15 Feb '12

Hello, Sorry to ask like that but could someone points me to some site that explains exactly what I need to open towards the internet so that my FS server is working while limiting its visibility? since 1st of February I have an IP that continually sends me SIP Register request at a rate of 70KB/s. I have complained to my internet provider but they refuse to help saying that the problem is on my side. I also logged a complain to the provider on that IP and am waiting on that. At the moment on my firewall I opened port 5060 and 5080 (well now I blocked as well that IP) but I want to know if both are really needed or if I could block one of them or may be limit the port to some IP. Any help/links will be gladly received thanks /Philippe

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Anyone have info on multicast paging FS and Linksys SPA phones
by Mario G 14 Feb '12

14 Feb '12

Scoured web sites, traces, etc. for a year no luck. Anyone out there with some info? Can't get paging to work on Linksys SPA962s, I have check the multicast address and it matches FS. Here is what happens: If I connect my old Linksys SPA9000 PBX and power it on, then turn it off, FreeSwitch paging works on all the phone! If a phone is powered off, FreeSwitch paging no longer works on that phone. Power on/off SPA9000 and all is well. Anyone out there have it working that could tell me the secret? I am using *11 to page to trigger the rtp_multicast_page extension. Thanks in advance! MarioG

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IVR + Cell Phone Weirdness
by George Sapak 14 Feb '12

14 Feb '12

I am having a weird issue with cell phones reaching the IVR, it seems the IVR only picks up the first number dialed not all three, this only happens from a cell. land lines are fine this is what I have to direct calls to internal extensions: <entry action="menu-exec-app" digits="/(^\*\d{3}$|^\d{3}$)/" param="transfer $1 XML default"/> Any help is greatly appreciated, Thank You, George

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FreeSWITCH Crash - How to Troubleshoot
by Lloyd Aloysius 14 Feb '12

14 Feb '12

Hi All, I was running the following version for a long time and never have any issue *freeswitch@internal> version FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)* * * *---------------------* * * Two days ago Brian suggest to update to a latest git for call pickup to working. So I did update. My New version is as follow * * *FreeSWITCH Version 1.0.head (git-eeeaf41 2012-02-10 17-27-50 -0600)* Today morning FreeSWITCH crash. How to troubleshoot this problem. How to find the reason .. why this crash? Thanks and regards, LLoyd

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SDP with 2 m=audio lines
by Sergey Okhapkin 14 Feb '12

14 Feb '12

I noticed that recent freeswitch versions could send SDP with 2 m=audio lines. Per RFC 3388, the behavior is undefined this case. I don't know how to reproduce the problem, I see such packets coming from my customers. Here is an example: INVITE sip:xxxxxxxxxxxx@sip.callwithus.com SIP/2.0 ... User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-eeeaf41 2012-02-10 17-27-50 -0600 .... v=0 o=FreeSWITCH 1329167961 1329167962 IN IP4 xx.xx.xx.xx s=FreeSWITCH c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 24598 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 m=audio 24598 RTP/AVP 98 0 8 3 101 13 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1; usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Hmm, I just realized that this SDP is buggy - 2 audio streams on the same port, with common codecs, but with different ptime :-)

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Inline execution concept
by virendra bhati 14 Feb '12

14 Feb '12

Hi List, I am the new guy for FreeSwitch. I am not clear what is the used of Inline execution. As per doing some tweak I found that Inline is used for set any variable for channel base like in asterisk we used double underscore( __ ). Am i right on this concept or not? Another question is how to include context in dialplan in XML ? I am reading FreeSwitch 1.0.6 for more details but still these question is not clear for me. Please help me to find out the answer of these questions. -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2

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segmentation fault on startup
by Darcy 14 Feb '12

14 Feb '12

I have three freeswitches running on public IPs, not behind a router. There was a power hit on site and all servers were restarted but none of the freeswitches started, when I tried to manually start them, I got a segmentation fault. I had to rename core.db on each system to get them to restart. Is this a known issue. One of these switches had a git update 1 week ago, the other two are various vintages but run flawless, until last night of course. The longest running one is about 14 months. Any input or thoughts would be appreciated. Darcy

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bypass media only if on same subnet?
by Tim St. Pierre 14 Feb '12

14 Feb '12

Hello, In order to eliminate NAT issues, we have built routable tunnels to our larger customers so that their phones are on a private subnet that is routable to a private subnet at our datacenter. Each Freeswitch system has a profile called nonat that uses an interface and address bound to this network. The remaining customers register to the "internal" profile, which is bound to a public IP address. At the moment, everything works just great - no nat issues, instant failover between the primary and secondary (shared registrations in DB), but all media flows through our network. If someone in office A calls another phone in office A, I would like FS to instruct the phones to send their media direct. The addresses and ports are all correct in this case. If someone in office A calls a phone in office B, I want Freeswitch to stay in the media path, as these two offices are not routable to each other, even though each is routable to Freeswitch. Is there a way to set up a profile (or dialplan) so that FS will bypass media only if the two endpoints are on the same subnet? An ACL isn't really the right thing, since it would require an exponential number of ACLs. Also, many calls go to ring groups, where several phones ring and we don't know which one will answer until it actually does (eliminating some sort of dialplan code using the rtp variables). Is this possible?

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questions about modules.conf
by Josh 14 Feb '12

14 Feb '12

According to the Installation/Build Guide (http://wiki.freeswitch.org/wiki/Installation_Guide) modules.conf gets generated during the execution of bootstrap.sh. Should I assume that the options specified in this file - both in terms of what modules are included as well as whether or not they are enabled - depends on my own configuration (the presence of required packages & libraries) as well as the presence of the source code/makefile for a particular module or is the content of this file "predefined" and filled in with a set of "standard" modules regardless of what bootstrap has found (I hope the answer is the former, but need to make absolutely sure)? Also, the same wiki page recommends that I also fetch freeswitch-contrib.git as well as freeswitch-sample-configs.git, but it is not clear to me whether I should dump the contents of these two projects within the main FS tree, where freeswitch.git was placed, or whether I should compile/build these separately - in a completely separate tree? Thanks!

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